[asterisk-dev] SIP <=> IAX - and RTP

[Darren Sessions]

I'm curious to know what the difference would be in how an RTP stream  
in a SIP vs. IAX call is handled internally in Asterisk. I've got an  
app_swift user that is complaining the app_swift module is producing  
garbled audio on his IAX channels, and perfect audio on the SIP  
channels. I was under the impression there wouldn't be a difference on  
the RTP side.

Any ideas before I tell him to troubleshoot his IAX setup?

Thanks!

_____________________________

Darren Sessions
dmsessions at gmail.com
http://www.darrensessions.com
_____________________________





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